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1070 commits

Author SHA1 Message Date
Sebastian Dröge
dc059efa60 rtp: Use GST_BUFFER_PTS() instead of GST_BUFFER_TIMESTAMP()
The mix between all these in the RTP code is confusing, let's try to be
consistent.
2015-06-10 14:34:47 +02:00
Ilya Konstantinov
c7e168ec70 rtpmanager: clarify negative lost packets in stats
Also:
- Move notes on units before field documentation.
- Unify documentation style.

https://bugzilla.gnome.org/show_bug.cgi?id=750653
2015-06-10 14:10:52 +02:00
Ilya Konstantinov
0a578c235a rtpmanager: document units of stats and arguments
Also, minor spelling and style corrections.

https://bugzilla.gnome.org/show_bug.cgi?id=750653
2015-06-09 18:21:59 +02:00
Sebastian Dröge
b549ebd066 rtpsession: Override the SSRC from the packets' SSRC if none was given via caps or property 2015-06-07 10:33:27 +02:00
Sebastian Dröge
d650a310da rtpsession: Only suggest our internal ssrc if it's not a random one and was selected as internal ssrc
https://bugzilla.gnome.org/show_bug.cgi?id=749581
2015-06-05 16:45:54 +02:00
Sebastian Dröge
8f5bdf9690 rtpjitterbuffer: Add support for receiving reduced size RTCP
It worked before but gave warnings, now we just ignore RTCP
packets that don't start with a SR. As all we're interested
in here are SRs.
2015-06-05 10:33:11 +02:00
Jose Antonio Santos Cadenas
f563176349 rtpssrcdemux: Add support for reduce size rtcp
According to RFC 5506, reduce size packages can be sent, this
packages may not be compound, so we need to add support for
getting ssrc from other types of packages.

https://bugzilla.gnome.org/show_bug.cgi?id=750327
2015-06-05 10:30:15 +02:00
Jose Antonio Santos Cadenas
f8f23bbf5d rtpsession: Add support for receiving reduced size rtcp
See RFC 5506

https://bugzilla.gnome.org/show_bug.cgi?id=750332
2015-06-05 10:24:17 +02:00
Sebastian Dröge
647eefea67 rtpsession: Only schedule a timer when we actually have to send RTCP
Otherwise we will have 10s-100s of thread wakeups in feedback profiles, create
RTCP packets, etc. just to suppress them in 99% of the cases (i.e. if no
feedback is actually pending and no regular RTCP has to be sent).

This improves CPU usage and battery life quite a lot.

https://bugzilla.gnome.org/show_bug.cgi?id=746543
2015-06-02 11:38:15 +02:00
Sebastian Dröge
8ada98964d rtpsession: Remove useless goto
https://bugzilla.gnome.org/show_bug.cgi?id=746543
2015-06-02 11:38:15 +02:00
Sebastian Dröge
506a8a8857 rtpbin: Add rtp-profile property for setting the default profile of newly created sessions
https://bugzilla.gnome.org/show_bug.cgi?id=746543
2015-06-02 11:38:15 +02:00
Sebastian Dröge
0f7e80ed59 rtpsession: Only put RRs and full SDES into regular RTCP packets
If we may suppress the packet due to the rules of RFC4585 (i.e. when
below the t-rr-int), we can send a smaller RTCP packet without RRs
and full SDES. In theory we could even send a minimal RTCP packet
according to RFC5506, but we don't support that yet.

https://bugzilla.gnome.org/show_bug.cgi?id=746543
2015-06-02 11:38:15 +02:00
Sebastian Dröge
6f830e5bd5 rtpsession: Keep track of tp/tn and t_rr_last separately
Otherwise we can't properly schedule RTCP in feedback profiles as we need to
distinguish the time when we last checked for sending RTCP (tp) but might have
suppressed it, and the time when we last actually sent a non-early RTCP
packet.

This together with the other changes should now properly implement RTCP
scheduling according to RFC4585, and especially allow us to send feedback
packets a lot if needed but only send regular RTCP packets every once in a
while.

https://bugzilla.gnome.org/show_bug.cgi?id=746543
2015-06-02 11:38:15 +02:00
Sebastian Dröge
3122ef4ae3 rtpsession: Add property for selecting RTP profile (AVP/AVPF/etc)
And modify our RTCP scheduling algorithm accordingly. We now can send more
RTCP packets if needed for feedback, but will throttle full RTCP packets by
rtcp-min-interval (t-rr-int from RFC4585).

In non-feedback mode, rtcp-min-interval is Tmin from RFC3550, which is
statically set to 1s or 0s by RFC4585. Tmin defines how often we should
send RTCP packets at most.

https://bugzilla.gnome.org/show_bug.cgi?id=746543
2015-06-02 11:38:15 +02:00
Sebastian Dröge
565cd49643 rtpsession: Don't crash if we receive FIR/PLI from a source we don't know 2015-05-21 13:26:53 +03:00
Santiago Carot-Nemesio
2fb1fe2ee3 rtpsession: Fix collection of statistics
Stats should be collected on the media rtp source not in the
sender one.

https://bugzilla.gnome.org/show_bug.cgi?id=749669
2015-05-21 12:56:12 +03:00
Sebastian Dröge
c60038f188 rtpsource: Queue bad packets instead of dropping them
So we can send them out once we found the next, consecutive sequence number in
case one is following.
2015-05-18 18:43:16 +03:00
Sebastian Dröge
9f18a271f3 rtpsource: Use g_queue_foreach() to unref all buffers in queues 2015-05-18 18:43:16 +03:00
Sebastian Dröge
54e924332e rtpsource: Refactor seqnum comparison code a bit 2015-05-18 18:43:16 +03:00
Sebastian Dröge
1974b24ef4 rtpsource: Allow sequence number wraparound during probation 2015-05-18 18:43:16 +03:00
Sebastian Dröge
3386de7a8a rtpsource: Make sequence number comparison code more readable
... by using gst_rtp_buffer_compare_seqnum() and signed integers
instead of implictly using effects of integer over/underflows.
2015-05-18 18:43:16 +03:00
Sebastian Dröge
ca110fb0b8 rtpjitterbuffer: When detecting a huge seqnum gap, wait for 5 consecutive packets before resetting everything
It might just be a late retransmission or spurious packet from elsewhere, but
resetting everything would mean that we will cause a noticeable hickup. Let's
get some confidence first that the sequence numbers changed for whatever
reason.

https://bugzilla.gnome.org/show_bug.cgi?id=747922
2015-05-18 18:43:15 +03:00
Tim-Philipp Müller
2e412a447a docs: update example pipelines in element docs
Mostly gst-launch -> gst-launch-1.0
Use autovideosink/autoaudiosink more often.
Sprinkle some converters here and there.
2015-05-10 11:05:00 +01:00
Sebastian Dröge
27729a2960 Revert "rtpsession: Also report internal sources in on-new-ssrc and on-ssrc-active"
This reverts commit d22ec49632.

Application code might expect that it only gets external sources on those
signals, and get confused by this. If anything we would need to add new
signals.
2015-05-07 14:51:45 +02:00
Sebastian Dröge
d22ec49632 rtpsession: Also report internal sources in on-new-ssrc and on-ssrc-active
Without this it seems impossible for an application to easily get notified
about the internal ssrcs that are created, e.g. sender sources, and also
to know when they are active and produce RTCP packets.

https://bugzilla.gnome.org/show_bug.cgi?id=746747
2015-05-06 11:21:22 +02:00
Sebastian Dröge
9d22ad421b rtpsession: The stats min_interval is in seconds, not nanoseconds
We have to scale it to compare it against our clock times.
2015-05-04 14:12:07 +02:00
Sebastian Dröge
afe1d5a89f rtpsession: Only return TRUE if early feedback was requested already and it's early enough 2015-05-04 14:11:00 +02:00
Sebastian Dröge
73c0c2920f rtpstats: Average RTCP packet size is in bytes, bandwidths in bits
We need to convert the size to bits for our calculations.

https://bugzilla.gnome.org/show_bug.cgi?id=747863
2015-04-27 16:45:40 +02:00
Sebastian Dröge
475b1e607e rtpstats: Use the same lower limit for RTCP bandwidth to stop sending RTCP everywhere
https://bugzilla.gnome.org/show_bug.cgi?id=747863
2015-04-27 16:45:33 +02:00
Sebastian Dröge
7596ed91b8 rtpsession: Use bandwidth calculation by default instead of some arbitrary hardcoded value
https://bugzilla.gnome.org/show_bug.cgi?id=747863
2015-04-27 16:45:25 +02:00
Sebastian Dröge
928cd110bc rtpsession: Bandwidth is supposed to be in bits/s, not bytes/s
https://bugzilla.gnome.org/show_bug.cgi?id=747863
2015-04-27 16:45:14 +02:00
Luis de Bethencourt
9391622579 Rename property enums from ARG_ to PROP_
Property enum items should be named PROP_ for consistency and readability.
2015-04-27 11:22:11 +01:00
Ilya Konstantinov
fd391a5404 rtpjitterbuffer: Fix "stats" property docs
https://bugzilla.gnome.org/show_bug.cgi?id=748436
2015-04-26 21:15:44 +02:00
Tim-Philipp Müller
d753a3eeb1 Remove obsolete Android build cruft
This is not needed any longer.
2015-04-26 17:55:07 +01:00
Luis de Bethencourt
671b4d25cd remove unused enum items PROP_LAST
This were probably added to the enums due to cargo cult programming and are
unused. Removing them.
2015-04-24 17:01:12 +01:00
Sebastian Dröge
edcc5be297 rtpjitterbuffer: When request retransmissions for future packets, consider the packet spacing in the extra delay
We now take the maximum of 2*jitter and 0.5*packet_spacing for the extra
delay. If jitter is very low, this should prevent unnecessary retransmission
requests to some degree.

https://bugzilla.gnome.org/show_bug.cgi?id=748041
2015-04-22 20:27:18 +02:00
Sebastian Dröge
3fe8ceff14 rtpjitterbuffer: Take a running average of the packet spacings instead of just the latest
https://bugzilla.gnome.org/show_bug.cgi?id=748041
2015-04-22 20:25:43 +02:00
Miguel París Díaz
f81c9a9568 rtpjitterbuffer: Add "rtx-next-seqnum" property
If this is set to FALSE, rtpjitterbuffer will not request retransmissions for
future packets based on when they are estimated to arrive.

See also https://bugzilla.gnome.org/show_bug.cgi?id=748041

https://bugzilla.gnome.org/show_bug.cgi?id=739868
2015-04-22 19:51:18 +02:00
Sebastian Dröge
68dfe93463 rtxreceive: Put debug output for retransmission requests at the right place
Before it was only ever printed once for every time a ssrc was associated with
a specific stream.
2015-04-22 19:51:18 +02:00
Sebastian Dröge
80268e7d37 rtpsource/rtprtxsend: Also pass correct seqnum-offset and payload to the RTX rtpsource
https://bugzilla.gnome.org/show_bug.cgi?id=747394
2015-04-16 17:33:37 +02:00
Arun Raghavan
26bec72e52 rtpsession: Track RTX ssrc caps
This is needed so that we can generate SR for RTX stream correctly (the
clock rate is required).

https://bugzilla.gnome.org/show_bug.cgi?id=747394
2015-04-16 17:33:37 +02:00
Sebastian Dröge
17c6532b75 rtprtxsend: Copy over timestamps from the orignal buffers to the RTX buffers
https://bugzilla.gnome.org/show_bug.cgi?id=747394
2015-04-16 17:33:37 +02:00
Sebastian Dröge
caa255d2ed rtprtx*: Fix typos 2015-04-14 19:08:38 +02:00
Sebastian Dröge
bd19b08d6d rtpsession: Not sending early RTCP now because of dithering means we send it with the next compound packet 2015-04-14 18:42:44 +02:00
Sebastian Dröge
4223d0c114 rtpsession: Improve debug output a bit if we can't allow early feedback 2015-04-14 18:42:44 +02:00
Sebastian Dröge
6c27293ffe rtpjitterbuffer: Change resyncing GST_WARNING to GST_INFO
This also happens in the very beginning when we receive the first packet, a
warning would be very confusing here. In all places where we should warn about
this, we would've printed a warning already before.
2015-04-13 20:25:48 +02:00
Miguel París Díaz
c4bb6a098b rtpjitterbuffer: Add "rtx-max-retries" property
This property allows to limit the maximum number of retransmission
for a specific packet.

https://bugzilla.gnome.org/show_bug.cgi?id=739868
2015-04-13 09:09:03 +02:00
Miguel París Díaz
05bd708fc5 rtpjitterbuffer: Fix expected_dts calc in calculate_expected
Right above we consider lost_packet packets, each of them having duration,
as lost and triggered their timers immediately. Below we use expected_dts
to schedule retransmission or schedule lost timers for the packets that
come after expected_dts.

As we just triggered lost_packets packets as lost, there's no point in
scheduling new timers for them and we can just skip over all lost packets.

https://bugzilla.gnome.org/show_bug.cgi?id=739868
2015-04-13 09:06:33 +02:00
Sebastian Dröge
1a2f253c3a rtpjitterbuffer: Make the next output buffer discont after resetting the jitterbuffer
Resetting the jitterbuffer drops all packets and other things, and will cause
a discontinuity in the packets received by the depayloaders. They should now
also flush anything they had pending as the new data will start at a different
position.

https://bugzilla.gnome.org/show_bug.cgi?id=739868
2015-04-13 09:05:34 +02:00
Tim-Philipp Müller
2fde2011b2 docs: make GstRTCPSync enum show up in rtpbin docs
https://bugzilla.gnome.org/show_bug.cgi?id=747358
2015-04-05 20:07:19 +01:00